Making FakeNetworkPipe demux audio and video packets.

webrtc/audio/test/low_bandwidth_audio_test.cc

Index: webrtc/audio/test/low_bandwidth_audio_test.cc
diff --git a/webrtc/audio/test/low_bandwidth_audio_test.cc b/webrtc/audio/test/low_bandwidth_audio_test.cc
index de955f75e19c936b98e61692f832d62254d8ffcc..65f28fa6a3385ccc2c30dbbac9ad332910202f54 100644
--- a/webrtc/audio/test/low_bandwidth_audio_test.cc
+++ b/webrtc/audio/test/low_bandwidth_audio_test.cc
@@ -20,12 +20,9 @@ namespace {
 // Wait half a second between stopping sending and stopping receiving audio.
 constexpr int kExtraRecordTimeMs = 500;
 
-// Large bitrate by default.
-const webrtc::CodecInst kDefaultCodec{120, "OPUS", 48000, 960, 2, 64000};
-

[oprypin_webrtc, 2017/04/06 11:45:32]

Why does the constant need to be moved?

[minyue-webrtc, 2017/04/06 11:56:33]

Because I do not want to use "120" any longer. (But yes, you may argue that we
can move line 92 in PS3 here. I see only one line of using the const, so I think
inline is better.)

I have a comment in PS3. The basic idea is that for tests derived from CallTest,
we try not to introduce new payload types than those defined in CallTest, so
that when one creates a Transport, he/she does not need to consider what
payload_type_map to give. Arbitrary payload types may also lead to conflicts.

 // The best that can be done with PESQ.
 constexpr int kAudioFileBitRate = 16000;
-}
+}  // namespace
 
 namespace webrtc {
 namespace test {
@@ -79,20 +76,20 @@ test::PacketTransport* AudioQualityTest::CreateSendTransport(
     Call* sender_call) {
   return new test::PacketTransport(
       sender_call, this, test::PacketTransport::kSender,
-      MediaType::AUDIO,
-      GetNetworkPipeConfig());
+      test::CallTest::payload_type_map_, GetNetworkPipeConfig());
 }
 
 test::PacketTransport* AudioQualityTest::CreateReceiveTransport() {
-  return new test::PacketTransport(nullptr, this,
-      test::PacketTransport::kReceiver, MediaType::AUDIO,
-      GetNetworkPipeConfig());
+  return new test::PacketTransport(
+      nullptr, this, test::PacketTransport::kReceiver,
+      test::CallTest::payload_type_map_, GetNetworkPipeConfig());
 }
 
 void AudioQualityTest::ModifyAudioConfigs(
   AudioSendStream::Config* send_config,
   std::vector<AudioReceiveStream::Config>* receive_configs) {
-  send_config->send_codec_spec.codec_inst = kDefaultCodec;
+  send_config->send_codec_spec.codec_inst = webrtc::CodecInst{
+      test::CallTest::kAudioSendPayloadType, "OPUS", 48000, 960, 2, 64000};

[minyue-webrtc, 2017/04/06 11:33:38]

use the payload defined in CallTest so that
CreateReceiveTransport/CreateSendTransport can use
test::CallTest::payload_type_map_

 }
 
 void AudioQualityTest::PerformTest() {
@@ -125,12 +122,12 @@ class Mobile2GNetworkTest : public AudioQualityTest {
   void ModifyAudioConfigs(AudioSendStream::Config* send_config,
       std::vector<AudioReceiveStream::Config>* receive_configs) override {
     send_config->send_codec_spec.codec_inst = CodecInst{
-        120,     // pltype
-        "OPUS",  // plname
-        48000,   // plfreq
-        2880,    // pacsize
-        1,       // channels
-        6000     // rate bits/sec
+        test::CallTest::kAudioSendPayloadType,  // pltype
+        "OPUS",                                 // plname
+        48000,                                  // plfreq
+        2880,                                   // pacsize
+        1,                                      // channels
+        6000                                    // rate bits/sec
     };
   }